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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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41 #include "talk/app/webrtc/webrtcsession.h" | 41 #include "talk/app/webrtc/webrtcsession.h" |
42 #include "webrtc/base/scoped_ptr.h" | 42 #include "webrtc/base/scoped_ptr.h" |
43 | 43 |
44 namespace webrtc { | 44 namespace webrtc { |
45 | 45 |
46 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration> | 46 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration> |
47 StunConfigurations; | 47 StunConfigurations; |
48 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration> | 48 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration> |
49 TurnConfigurations; | 49 TurnConfigurations; |
50 | 50 |
| 51 // Parses the URLs for each server in |servers| to build |stun_config| and |
| 52 // |turn_config|. |
| 53 bool ParseIceServers(const PeerConnectionInterface::IceServers& servers, |
| 54 StunConfigurations* stun_config, |
| 55 TurnConfigurations* turn_config); |
| 56 |
51 // PeerConnection implements the PeerConnectionInterface interface. | 57 // PeerConnection implements the PeerConnectionInterface interface. |
52 // It uses MediaStreamSignaling and WebRtcSession to implement | 58 // It uses MediaStreamSignaling and WebRtcSession to implement |
53 // the PeerConnection functionality. | 59 // the PeerConnection functionality. |
54 class PeerConnection : public PeerConnectionInterface, | 60 class PeerConnection : public PeerConnectionInterface, |
55 public MediaStreamSignalingObserver, | 61 public MediaStreamSignalingObserver, |
56 public IceObserver, | 62 public IceObserver, |
57 public rtc::MessageHandler, | 63 public rtc::MessageHandler, |
58 public sigslot::has_slots<> { | 64 public sigslot::has_slots<> { |
59 public: | 65 public: |
60 explicit PeerConnection(PeerConnectionFactory* factory); | 66 explicit PeerConnection(PeerConnectionFactory* factory); |
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196 rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_; | 202 rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_; |
197 rtc::scoped_ptr<StatsCollector> stats_; | 203 rtc::scoped_ptr<StatsCollector> stats_; |
198 | 204 |
199 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_; | 205 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_; |
200 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_; | 206 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_; |
201 }; | 207 }; |
202 | 208 |
203 } // namespace webrtc | 209 } // namespace webrtc |
204 | 210 |
205 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ | 211 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ |
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